CN1055585C - Code exciting lnear predict coder and decoder - Google Patents
Code exciting lnear predict coder and decoder Download PDFInfo
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- G10L13/00—Speech synthesis; Text to speech systems
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- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
- G10L19/26—Pre-filtering or post-filtering
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- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
- G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
- G10L19/10—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a multipulse excitation
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- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
- G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
- G10L19/12—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders
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- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
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- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
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- G10L25/03—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the type of extracted parameters
- G10L25/24—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the type of extracted parameters the extracted parameters being the cepstrum
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Abstract
The invention is directed to obtain a reproduced sound signal of high quality of a low encoding speed. The sound channel prediction signal contained in the encoding sound signal is taken as LSP coefficient. Selectively using noise code book or pulse code book, the frequency characteristic of primary sound signal is reflected by an optimum excitation signal. Thus, the gain-control of self-adapting excitation signal and optimum excitation signal is carried out according to sound power.
Description
The present invention relates to encoder and decoder, for example, go for having the telephone set of so-called absence recording function according to code exciting lnear predict (CE, LP) coded system.
In the telephone set with absence recording function, in the past, majority was to use the recording of information medium of cassette tape as metered call person or callee.
But when using cassette tape as recording medium, the recording of information revived structure will occupy a lot of spaces, in addition, when a plurality of information, exist find the beginning of wanting the information of listening need the cancellation of regular hour and information unit be difficulty etc. problem.
The scheme of use semiconductor memory (IC memory) therefore, has been proposed as the recording of information medium.Like this, when using the IC memory,, preferably use the compression coding mode that voice signal is compressed laggard line item, expands when regenerating if think and to write down a lot of information with simple as far as possible structure as carrier.
As is generally known the high efficiency of compression coded system as to voice signal has the code-excited linear prediction mode.The code-excited linear prediction mode is for the transmission of the narrow sense of voice signal proposes, and is at can the regenerate voice signal of faithful to as far as possible input audio signal of decoding one side with seldom conveying capacity.
But, in telephone set,, when using existing code-excited linear prediction mode, can not satisfy the various requirement that is associated with the absence recording function as the compression coding mode of recording of information regeneration usefulness with absence recording function.
Therefore, for inapplicable devices such as telephone set of code-excited linear prediction mode so far, seeking suitable code exciting lnear predict coder and decoder in order to be suitable for absence recording function.
In order to address the above problem, in the 1st the present invention, use following each several part to constitute code exciting lnear predict coder.
That is, forming device by (1) channel information maker, (2) sound power quantizer, (3) self adaptation code table, (4) noise code table, (5) pulse code table, (6) constant excitation signal selector, (7) frequency characteristic operator, (8) gain code table, (9) adder, (10) Optimum Excitation signal searcher of Fig and (11) coded sound signal constitutes.The channel information maker obtains the LPC coefficient, quantizes after it is transformed to the LSP coefficient according to the synthetic video signal of original sound signal or local regeneration, and the LSP coefficient that will quantize simultaneously carries out inverse quantization, reverts to the LPC coefficient; The sound power quantizer calculates the power of original sound signal and quantizes and the inverse quantization processing; The self adaptation code table is exported the adaptive excitation signal of renewal adaptively; Noise code table output noise pumping signal; Pulse code table output pulse feature pumping signal; The constant excitation signal selector is selected the pumping signal of noise code table and pulse code table; The frequency characteristic operator is transformed to the frequency characteristic that has with the original sound signal similar with the noise pumping signal or the pulse feature pumping signal of the output of constant excitation signal selector; Gain code table output with from the adaptive excitation signal of self adaptation code table output with from the noise pumping signal of frequency characteristic operator output or pulse feature pumping signal excitation gain corresponding, that determine by the inverse quantization power of sound power quantizer output; What adder will have been carried out gain controlling carries out add operation from the adaptive excitation signal of self adaptation code table output with from the noise pumping signal or the pulse feature pumping signal of the output of frequency characteristic operator, forms final pumping signal; The Optimum Excitation signal searcher of Fig determines to use the output drive signal that also is to use which code table noise code table and the pulse code table from the Optimum Excitation signal of self adaptation code table, noise code table, pulse code table and the output of gain code table according to from the LPC coefficient of channel information maker output and the final pumping signal of adder output; Code signal forms device when having determined the Optimum Excitation signal, forms coded sound signal according to the index of various code tables, the selection mode index of constant excitation signal selector, the quantized power of sound power quantizer and the quantification LSP coefficient of channel information maker.
In addition, the 2nd code exciting lnear predict decoder of the present invention is characterised in that: corresponding with the 1st code exciting lnear predict coder of the present invention, constitute by following each several part.
That is, form device by (1) coded sound signal separator, (2) channel information regenerator, (3) sound power inverse DCT, (4) self adaptation code table, (5) noise code table, (6) pulse code table, (7) constant excitation signal selector, (8) frequency characteristic operator, (9) gain code table, (10) adder and (11) regeneration voice signal.The coded sound signal separator is separated into various information with coded sound signal; The quantification LSP coefficient that the channel information regenerator will separate carries out inverse quantization to be handled, and reverts to the LPC coefficient; The sound power inverse DCT carries out inverse quantization with the quantized power of separating to be handled; Output of self adaptation code table and the corresponding adaptive excitation signal of adaptive excitation signal index that separates; The noise code table is exported the noise pumping signal corresponding with this index when having separated noise pumping signal index; The pulse code table is exported the pulse feature pumping signal corresponding with this index when having separated pulse feature pumping signal index; The constant excitation signal selector is selected the pumping signal of noise code table and pulse code table according to the selection mode index that separates; The frequency characteristic operator will be transformed to the frequency characteristic that has with the original sound signal similar from the noise pumping signal or the pulse feature pumping signal of this constant excitation signal selector output; Gain code table output with from the adaptive excitation signal of self adaptation code table output with from the noise pumping signal of frequency characteristic operator output or the pulse feature pumping signal is corresponding, by the excitation gain of determining from the inverse quantization power of sound power inverse DCT output and the excitation gain that separates; Adder will be carried out add operation from the adaptive excitation signal of the self adaptation code table output of carrying out gain controlling with from the noise pumping signal or the pulse feature pumping signal of the output of frequency characteristic operator, form final pumping signal; The regeneration voice signal forms device according to forming the regeneration voice signal from the LPC coefficient of channel information regenerator output with from the final pumping signal that adder is exported.
The 3rd of the present invention is characterised in that: in the code exciting lnear predict coder that has the self adaptation code table at least, has the exponential transform device, when having indicated the tone control pattern, this exponential transform device will be according to the definite adaptive excitation signal index of self adaptation code table, be transformed to fixing adaptive excitation signal index, and supply with coded sound signal formation device so that contain fixing adaptive excitation signal index in the coded sound signal.
The 4th of the present invention is characterised in that: in having the code exciting lnear predict decoder of self adaptation code table at least, has the exponential transform device, when having indicated the tone control pattern, this exponential transform device will be transformed to fixing adaptive excitation signal index and supply with the self adaptation code table according to the definite adaptive excitation signal index of self adaptation code table that obtains after coded sound signal is separated.
The 5th of the present invention is characterised in that: in the code exciting lnear predict coder of the original sound signal of input being encoded according to the code-excited linear prediction mode, has the reproduction speed controller, when having indicated the reproduction speed control model, take out between this reproduction speed controller carries out the original sound signal according to the change multiplying power of indication and the periodicity that has of original sound signal or interpolation is handled, and supply with encode processor.
The 6th of the present invention is characterised in that: have the pumping signal regenerator that forms pumping signal according to the coded sound signal of input, form the channel information regenerator of sound channel predictive coefficient according to the coded sound signal of input, the regeneration voice signal that forms the regeneration voice signal with sound channel predictive coefficient according to the pumping signal of pumping signal regenerator and sound channel information regeneration device forms in the code exciting lnear predict decoder of device, has the reproduction speed controller, when having indicated the reproduction speed control model, this reproduction speed controller is according to the change multiplying power and the periodicity that has of pumping signal of indication, takes out between the pumping signal of pumping signal regenerator is carried out or interpolation is handled and supplied with the regeneration voice signal and forms device.
The 7th of the present invention is characterised in that: have the pumping signal regenerator that forms pumping signal according to the coded sound signal of input, form the channel information regenerator of sound channel predictive coefficient according to the coded sound signal of input, the regeneration voice signal that forms the regeneration voice signal with sound channel predictive coefficient according to the pumping signal of pumping signal regenerator and sound channel information regeneration device forms in the code exciting lnear predict decoder of device, and the back level that forms device at the regeneration voice signal is provided with the white noise adder of adding white noise in the regeneration voice signal.
The 1st code exciting lnear predict coder of the present invention and the 2nd code exciting lnear predict decoder of the present invention are intended to realize low coding rate, have symmetrical structure.Therefore, conclude its effect of explanation below.
As the sound channel predictive coefficient that comprises in the coded sound signal, use the LSP coefficient.It is good that its reason is the interpolation characteristic of the frequency characteristic of sound channel, even the LSP coefficient utilizes coding figure place seldom to encode, also causes that than LPC coefficient etc. the distortion of vocal tract spectrum is little, by with the vector quantization method combination, can carry out high efficiency coding.
In addition, as code table, except general self adaptation code table and noise code off-balancesheet, also be provided with pulse code table and gain code table, use noise code table and pulse code table that selected noise pumping signal or pulse feature pumping signal are operated selectively, can have frequency characteristic with the original sound signal similar.
Why use noise code table and pulse code table selectively, be contribution, even and use the pulse feature pumping signal also can adapt to low coding rate to the clear and definite steady component that sound is arranged of the periodic very strong forward position that sound is arranged and pulse feature.That is, even under the few situation of the figure place of distributing, also can form good pumping signal.Listen to carry out the frequency characteristic operation, though the frequency characteristic that is pumping signal is in theory as white and medelling, but, in fact be not white, have the characteristic close with the frequency characteristic of original sound signal, as long as make the frequency characteristic of the frequency characteristic of noise pumping signal and pulse feature pumping signal and original sound signal close, just can obtain high-quality regeneration voice signal, in addition, the effective frequency composition of pumping signal is far longer than quantization error signal, can obtain the masking effect of quantization error signal.Why carry out gain controlling, be under the big situation of the restriction of coding figure place, though can not make the length of pumping signal big,, in this case,, can realize high-quality by introducing the gain controlling corresponding with sound power.
For the above reasons, in the 2nd code exciting lnear predict decoder of the 1st code exciting lnear predict coder of the present invention and invention, utilize low coding rate coded sound signal can obtain high-quality regeneration voice signal.
The 3rd code exciting lnear predict coder of the present invention and the 4th code exciting lnear predict decoder of the present invention all make the tone of regeneration voice signal variable, the former does not change decoder one side, utilize the processing of encoder one side to make the tone of regeneration voice signal variable, the latter does not change encoder one side, utilizes the processing of decoder one side to make the tone of regeneration voice signal variable.
From the index of the adaptive excitation signal of self adaptation code table output, get on very well roughly, be subjected to the having the greatest impact of frequency of voice signal, if index is fixed, the frequency (tone) of regeneration voice signal is fixed.Therefore, consider and do not give other each several parts with influence, in the 3rd code exciting lnear predict coder of the present invention, before the formation device of coded sound signal, in addition, in the 4th code exciting lnear predict decoder of the present invention, supply with the path of self adaptation code table at the index that will separate from coded sound signal, be provided with the exponential transform device, the adaptive excitation signal exponential transform that is used for determining according to the self adaptation code table is fixing adaptive excitation signal index.
The speed that the 5th code exciting lnear predict coder of the present invention and the 6th code exciting lnear predict decoder of the present invention can make the original sound signal have becomes the speed of regeneration voice signal, the former does not change decoder one side, utilize the processing of encoder one side to make the speed of regeneration voice signal variable, the latter does not change encoder one side, utilizes the processing of decoder one side to make the speed of regeneration voice signal variable.
In order to make reproduction speed variable, can be from taking out between sampling is carried out or the interpolation processing.As long as take out or interpolation between carrying out simply, just be easy to generate discontinuity point.Therefore, considering the periodicity of signal, can be to take out or interpolation between unit carries out with its cycle.Here, velocity transformation is not exerted an influence as far as possible to other structures, and the reduction of the quality of the voice signal of also wishing to regenerate is restricted to Min..Therefore, in the 5th code exciting lnear predict coder of the present invention, input stage at the original sound signal, perhaps, in the 6th code exciting lnear predict decoder of the present invention, form in the path of device pumping signal being supplied with the regeneration voice signal, be provided with the reproduction speed controller that periodicity that change multiplying power and process object signal according to indication have is taken out between the process object signal is carried out or interpolation is handled.
The 7th code exciting lnear predict decoder of the present invention is considered along with becoming noise composition in the low coding rate regeneration voice signal and is formed the microphonic noise easily and constitute (in this manual, to be modulated by white noise and become different with the white noise phenomenons that influence the tone color that ear listens and be called noise, will influence the noise that ear listens later on and be called the microphonic noise).If white noise is added in the microphonic noise, it is not obvious that the microphonic noise just becomes, and approach the voice signal of nature.Therefore, the 7th code exciting lnear predict decoder of the present invention is provided with the white noise adder that adds white noise in the regeneration voice signal in the back level of regeneration voice signal formation device.
Fig. 1 is the block diagram of the 1st embodiment of code exciting lnear predict coder.
Fig. 2 is the block diagram of the 1st embodiment of code exciting lnear predict decoder.
Fig. 3 is the block diagram of the 2nd embodiment of code exciting lnear predict coder.
Fig. 4 is the block diagram of the 2nd embodiment of code exciting lnear predict decoder.
Fig. 5 is the block diagram of the 3rd embodiment of code exciting lnear predict coder.
One of Fig. 6 takes out between being/the operating instruction figure of interpolation operator 132 ().
Fig. 7 takes out between being/the operating instruction figure of interpolation operator 132 (two).
Fig. 8 is the block diagram of the 3rd embodiment of code exciting lnear predict decoder.
Fig. 9 is the block diagram of the 4th embodiment of code exciting lnear predict decoder.
101... sound channel analyzer, 102... sound channel predictive coefficient quantizer,
103,213... composite filter, 104... frame power quantization device,
105,204... self adaptation code table, 106,205... noise code table,
107,206... pulse code table, 108,207... gain code table,
109,209... transform vector device,
110,111,210, the 211... multiplier
112,212, the 241... adder
113,208... constant excitation vector selector switch
114... Weighted distance calculator, 115... code table searcher
116,201... memory interface, 120,220... exponential transform device
130,230... buffer storage, 131,231... periodicity analysis device
132, take out between 232.../the interpolation operator
202... sound channel predictive coefficient inverse DCT
203... frame power inverse DCT, 214... postfilter
240... noise generator
[embodiment]
(A) the 1st embodiment of code exciting lnear predict coder
Fig. 1 is the 1st embodiment of code exciting lnear predict coder of the present invention, for example coded sound signal can be stored in the IC memory with the telephone set of absence recording function.
The 1st embodiment of the described code exciting lnear predict decoder of the 1st embodiment of this code exciting lnear predict coder and back is in order to be conceived to low coding rate (for example, 4kbit/s) with a lot of information stores in the IC memory.
In Fig. 1, from input terminal 100 frame by frame unit conclude original sound vector (original sound signal) the S incoming frame power quantization device 104 import as vector.Carry out quantification treatment behind the power of frame power quantization device 104 calculating original sound vector S, its index Io to memory interface 116 outputs, simultaneously, behind the calculating inverse quantization value P, is exported to gain code table 108.
In addition, original sound vector S input sound channel analyzer 101 calculates sound channel predictive coefficient (LPC coefficient) a, and transmits to sound channel predictive coefficient quantizer 102.Sound channel predictive coefficient quantizer 102 carries out quantification treatment after LPC coefficient a is transformed to LSP (Line Spectrum Pair) coefficient, and its index Ic is exported to memory interface 116.In addition, sound channel predictive coefficient quantizer 102 calculates the inverse quantization value of LSP coefficient according to index Ic, be transformed to LPC coefficient aq after, to composite filter 103 and 109 outputs of transform vector device.
Here, as the sound channel predictive coefficient that comprises in the record code (coded sound signal), why use the LSP coefficient to be, interpolation characteristic to the frequency characteristic of sound channel is good, even under coding figure place seldom, encode, the LSP coefficient also causes that than LPC coefficient etc. the distortion of vocal tract spectrum is little, by with the vector quantization method combination, can carry out high efficiency coding.
Search for the synthetic video vector S w of this part regeneration, find out the Optimum Excitation vector e that it approaches the original sound vector S most, the index of various code tables 105~108 at this moment etc. is included in the record code.
In addition, and for example ask the situation of sound channel predictive coefficient and frame power opposite by every frame, the search of the pumping signal e of the described the best in back is then undertaken by 1 frame is divided into a plurality of subframe units.
Under the situation of present embodiment,, be provided with self adaptation code table 105, noise code table 106, pulse code table 107 and gain code table 108 as code table.
Self adaptation code table 105, noise code table 106 and pulse code table 107 are being stored the waveform codes vector (as adaptive excitation vector, noise excitation vectors, the pulse feature excitation vectors of pumping signal) about pumping signal respectively, and gain code table 108 is being stored the gain code about adaptive excitation vector and constant excitation vector (with the address of noise excitation vectors and the merging of pulse feature excitation vectors).
Adaptive excitation vector and noise excitation vectors be respectively with the same in the past, be on the statistics to periodically strong the contributive waveform stimulus vector of sound is arranged and on statistics to the periodically weak contributive waveform stimulus vector of voiceless sound at random.In addition, the adaptive excitation vector of self adaptation code table 105 upgrading like that adaptively as hereinafter described.The pulse feature excitation vectors is the waveform stimulus vector that is made of isolated pulse.The pulse feature excitation vectors is the contribution of having considered the clear and definite steady component that sound is arranged of the periodically strong forward position that sound is arranged and pulse feature.Gain code is for example carried out the vector quantization processing, and the part of code is the code about the gain of adaptive excitation vector, and another part is about the code of the gain of constant excitation vector (two-dimentional quantization table).
In addition, the sound-source signal of pulse feature is to have a periodic simple signal, so, also can think what pulse signal generator took place, still, preferably as present embodiment, encoding then reads out generation from code table 107.Its reason is down, that is, output easy and self adaptation code table 105 is synchronous, in addition, by adopting the books structure identical with noise code table 106, as hereinafter described, the multichannel processing when concluding in the record code after selection noise excitation vectors or the pulse feature excitation vectors etc. are just easy.
Use so various excitation vectors, the synthetic video vector S w that obtains local regeneration is the most similar to the original sound vector S, behind the Optimum Excitation vector about various excitation vectors, its index is supplied with memory interface 116, bring in the record code (coded sound signal), store in the not shown IC memory.Because present embodiment is conceived to low coding rate, so, for the constant excitation vector, select noise excitation vectors or pulse feature excitation vectors and write down its index.Therefore, select some selection information to be also contained in the record code as the constant excitation vector.
The search of this Optimum Excitation vector (selection that comprises noise excitation vectors or pulse feature excitation vectors is handled) is supposed here according to the order of adaptive excitation vector, noise excitation vectors, pulse feature excitation vectors, gain code and is carried out and describe.In addition, as long as can access best adaptive excitation vector, noise excitation vectors, pulse feature excitation vectors and gain code, this search order etc. is not limited to the following description.
When carrying out the search of best endoadaptation excitation vectors, make noise code table 106 and pulse feature code table 107 be output as 0, in addition, multiplier 110 multiply by the multiplying of the gain coefficient bk (for example 1) of desired value.Under such state, export all adaptive excitation vector eai that are stored in the self adaptation code table 105 in chronological order or concurrently, be defeated by composite filter 103 as excitation vectors by multiplier 110 and adder 112.Composite filter 103 carries out process of convolution as braning factor to this excitation vectors ea (eai) with LPC coefficient aq, (i=1~n) asks the synthetic video vector (representing with Swi) of the content that only reflects adaptive excitation vector eai here, to all adaptive excitation vector eai as the sound source parameter.
The distance calculator 114 that cum rights is heavy carries out the subtraction of the synthetic video vector S wi of original sound vector S and each candidate, and then carries out after the weighting of frequency, and the vector of each candidate is calculated the quadratic sum ew (ewi) of each composition, and to 115 outputs of code table searcher.Code table searcher 115 will be corresponding with the minimum value among n the quadratic sum ewi the adaptive excitation vector ea of minimum be defined as best adaptive excitation vector.
Then, carry out the search of best noise excitation vectors.When carrying out this search, constant excitation vector selector switch 113 is switched to noise code table 106 1 sides, make self adaptation code table 105 be output as 0 (output optimal self-adaptive excitation vectors also can), in addition, multiplier 111 multiply by the multiplying of the gain coefficient gk (for example 1) of desired value.Under such state, according to time sequencing or concurrently output be stored in all noise excitation vectors esj in the noise code table 106 (j=1~m), and be input in the transform vector device (frequency characteristic operator) 109 by constant excitation vector selector switch 113.
After, carrying out the identical processing of search with best adaptive excitation vector, code table searcher 115 is determined best noise excitation vectors es.
Here, why transform vector device 109 is set, is following reason.In the past, the frequency characteristic of excitation vectors was in theory as white and medelling, still, in fact was not white, experimentally confirmed to have the characteristic close with the frequency characteristic of original sound vector S.Therefore, as long as make the frequency characteristic of the frequency characteristic of noise excitation vectors and pulse feature excitation vectors and original sound vector S approaching, just can obtain high-quality synthetic video vector like that, in addition, the effective frequency composition of excitation vectors is far longer than quantization error signal, thereby can obtain the masking effect of quantization error signal.Therefore, transform vector device 109 is set.Here, LPC coefficient ac is arranged, in addition, represent information (the comprising the gain corresponding) Ia of adaptive excitation vector of the best of tone information of forecasting in addition with it as the information of the frequency characteristic of expression original sound vector S.Therefore, transform vector device 109 carries out the operation of the frequency characteristic of noise excitation vectors and pulse feature excitation vectors according to these information.
Like this, when the search of the noise excitation vectors of the best finishes, carry out the search of best pulse feature excitation vectors with that.When carrying out this search, constant excitation vector selector switch 113 is switched to pulse feature code table 107 1 sides, make self adaptation code table 105 be output as 0 (also can export the optimal self-adaptive excitation vectors), in addition, multiplier 111 multiply by the multiplying of the gain coefficient gk (for example 1) of desired value.Under such state, according to time sequencing or concurrently output be stored in all pulse feature excitation vectors epk in the pulse feature code table 107 (k=1~m).Later processing and the same when carrying out the search of best noise excitation vectors is so explanation is omitted.
Like this, when having determined best pulse feature excitation vectors ep, code table searcher 115 compares the information of the constant excitation vector that is defined as writing down that quadratic sum ew is little with the quadratic sum ew of the quadratic sum ew of the noise excitation vectors es of the best and best pulse feature excitation vectors ep.
After this, carry out the search of best gain code.When carrying out the search of this gain code, from the best adaptive excitation vector ea of self adaptation code table 105 outputs, constant excitation vector selector switch 113 is switched to selected noise code table 106 or pulse feature code table 107 1 sides, from fixedly code table 106 or best constant excitation vector es or the ep of 107 outputs that selects.The gain that gain of being used by the adaptive excitation vector from 1 gain code of gain code table 108 output and constant excitation vector are used constitutes, after in these gains, reflecting the frame power P, (k=1~t) be defeated by multiplier 110, the gain gk that the constant excitation vector is used is defeated by multiplier 111 to the gain bk that the adaptive excitation vector is used.Like this, just carry out add operation, and be defeated by composite filter 103 as excitation vectors e by 112 pairs of adders optimal self-adaptive excitation vectors of carrying out gain controlling and the optimal fixation excitation vectors of carrying out operation of overfrequency characteristic and gain controlling.To all gain code in the gain code table 108 according to time sequencing or carry out such processing concurrently.Processing during the later search of the heavy composite filter of cum rights 103, the processing during with the search of carrying out various excitation vectors is the same.
When code table searcher 115 obtains optimal self-adaptive excitation vectors, optimal fixation excitation vectors and optimum gain code, just index Ia, Is or Ip and the Ig with them supplies with memory interface 116, simultaneously, also will represent to select which the fixed code selector switch information Iw in noise excitation vectors and the pulse feature excitation vectors to supply with memory interface 116.
In addition, the code table searcher 115 exponential sum fixed code selector switch information that will supply with memory interface 116 is defeated by corresponding code table (105 and 108,106 or 107) and fixed code selector switch 113.At this moment, switch 113 switches, from each code table output Optimum Excitation vector and optimum code.Like this, just when the frame that carries out this is handled, also can form the excitation vectors e (eo) of the synthetic video vector S w that approaches the original sound vector S most, and it is supplied with self adaptation code table 105 from adder 112 outputs.And self adaptation code table 105 carries out the renewal of adaptive excitation vector C ai to be handled.
Above encoding process is carried out repeatedly to every frame and subframe, and coded sound signal M journal is in the IC memory.
In addition, telephone set for band absence recording function, when the information of the record message of its owner (callee) when going out and the metered call person when user's information conveyed of going out, such encoding process is carried out according to the instruction of the controller (CPU) of the whole telephone set of control.
Therefore,, under low coding rate, also can obtain high-quality regeneration sound according to the code exciting lnear predict coder of above-mentioned the 1st embodiment, can be in the IC memory with a large amount of information stores.
Below, specifically describe under low coding rate and also can obtain high-quality regeneration sound.
(1) when coding rate is hanged down in employing, because it is few to distribute to the coding figure place of sound source parameter (pumping signal), so, the constant excitation vector of preparing is also few, the pulse feature noise that is difficult to regenerate clearly and comprises in the original sound vector S is under the situation of present embodiment, owing to utilized the pulse feature excitation vectors, so, can improve the regeneration quality of sound at this moment.
In addition, because after switch pulse excitation vectors and noise excitation vectors, use, so, can be corresponding with low coding rate, simultaneously, can improve the regeneration quality that object has the transition portion of sound signal such, random signal and pulse to mix the signal that exists.
(2) when coding rate was hanged down in employing, not only the coding figure place to the sound source parameter was few, and also few to the coding figure place of sound channel parameter.Under the situation of present embodiment, even encode, also can write down the information of the little LSP coefficient of the vocal tract spectrum distortion that causes than LPC coefficient etc. with few coding figure place, so, can improve the regeneration quality.
(3) as described above, owing to considering that actual pumping signal (corresponding excitation vectors e) has the frequency characteristic close with the frequency characteristic of input audio signal (corresponding original sound vector S) and is provided with transform vector device 109, so, in fact can correspondingly improve the regeneration quality, simultaneously, have masking effect, thereby can improve the regeneration quality the quantization error signal of following this conversion.
(B) the 1st embodiment of code exciting lnear predict decoder
Below, the 1st embodiment of the code exciting lnear predict decoder that present invention will be described in detail with reference to the accompanying.Present embodiment is corresponding with the 1st embodiment of code exciting lnear predict coder shown in Figure 1, has the structure shown in the block diagram of Fig. 2.
In Fig. 2, the code exciting lnear predict decoder of the 1st embodiment is made of memory interface 201, sound channel predictive coefficient inverse DCT 202, frame power inverse DCT 203, self adaptation code table 204, noise code table 205, pulse code table 206, gain code table 207, constant excitation vector selector switch 208, transform vector device 209, multiplier 210,211, adder 212, composite filter 213 and postfilter 214.
Read and from the coded sound signal M of these code exciting lnear predict decoders of input terminal 200 input, input store interface 201 from the IC memory.Memory interface 201 is separated into this coded sound signal M the index Ig and the constant excitation vector selector switch information Iw of the index Is of index Ia, the optimal fixation excitation vectors es of quantitative information Ic, frame power information Io, optimal self-adaptive excitation vectors ea of LSP coefficient or ep or Ip, optimum gain code.And, the quantitative information Ic of LSP coefficient is supplied with sound channel predictive coefficient inverse DCT 202, frame power information Io is supplied with frame power inverse DCT 203, the index Ia of optimal self-adaptive excitation vectors ea is supplied with self adaptation code table 204 and transform vector device 209, the index Ig of optimum gain code is supplied with gain code table 207, constant excitation vector selector switch information Iw is supplied with constant excitation vector selector switch 208.In addition, index Is or the Ip with optimal fixation excitation vectors es or ep supplies with noise code table 205 or the pulse code table of determining according to constant excitation vector selector switch information Iw 206.
The LSP coefficient through coding that sound channel predictive coefficient inverse DCT 202 will be supplied with is deciphered (for example, carry out the vector inverse quantization and handle), is transformed to LPC coefficient aq again.The LPC coefficient aq of conversion like this is as sound channel predictive coefficient information providing transform vector device 209, composite filter 213 and postfilter 214.
Frame power inverse DCT 203 is asked frame power inverse quantization value (the frame power of regeneration) P according to frame power information Io, and supplies with gain code table 207.
Gain code table 207 with the frame power P the adaptive excitation vector of determining by the index Ig that supplies with and the gain code used of constant excitation vector in reflect the multiplier 211 used of multiplier 210 that gain code b, g supply adaptive excitation vector is used and constant excitation vector respectively.
The adaptive excitation vector ea that 204 outputs of self adaptation code table are determined by the index Ia that supplies with, this adaptive excitation vector ea carries out gain controlling by multiplier 210, and supplies with adder 212.
Noise code table 205 or pulse code table 206 are exported to transform vector device 209 by constant excitation vector selector switch 208 with noise excitation vectors es or the pulse feature excitation vectors ep corresponding with index Is that supplies with or Ip, and transform vector device 209 carries out its frequency characteristic operation according to the index Is of LPC coefficient aq, adaptive excitation vector es.Like this, the constant excitation vector ev through the frequency characteristic operation is undertaken being defeated by adder 212 after the gain controlling by gain controller 211.
In addition, the excitation vectors e from adder 212 outputs also is defeated by self adaptation code table 204.At this moment, self adaptation code table 204 uses this excitation vectors e to carry out the renewal of adaptive excitation vector.
The code exciting lnear predict decoder is each when supplying with coded sound signal, promptly every frame (for pumping signal exactly to every subframe) carried out above-mentioned processing.
Therefore, code exciting lnear predict decoder according to the 1st embodiment, owing to have the structure of handling the LSP coefficient of supplying with, as sound source pulse code table 206 is arranged, have the frequency characteristic that makes stationary sound source and the approaching transform vector device 209 of frequency characteristic of input audio signal, so, just can make the effect of code exciting lnear predict coder of above-mentioned the 1st embodiment actual effectively in view of the above.
(C) the 2nd embodiment of code exciting lnear predict coder
Fig. 3 is the 2nd embodiment of code exciting lnear predict coder of the present invention, is the example that for example coded sound signal can be stored into in the IC memory of the telephone set of absence recording function.In Fig. 3, be marked with identical symbol for the part identical and corresponding with Fig. 1.
The code exciting lnear predict coder of the 2nd embodiment is so that be prerequisite with the code exciting lnear predict decoder of above-mentioned the 1st embodiment as decoder.
The code exciting lnear predict decoder of described the 2nd embodiment in the code exciting lnear predict coder of the 2nd embodiment and back is conceived to make the periodicity (height of sound) of regeneration sound vector to keep certain.
According to Fig. 3 and Fig. 1 more as can be known, the code exciting lnear predict coder of the 2nd embodiment is to have increased exponential transform device 120 in the structure of the 1st embodiment.The index Ia of tone control signal con1 and optimal self-adaptive excitation vectors ea imports this exponential transform device 120.When tone control signal con1 indicates the non-control state of tone, exponential transform device 120 just makes the index Ia of optimal self-adaptive excitation vectors ea directly pass through, supply with memory interface 116, when tone control signal con1 indicates the state of a control of tone, the index Iac that irrespectively fixes with the index Ia of optimal self-adaptive excitation vectors ea just, and supply with memory interface 116.
Here, the index Ia of self adaptation code table 105 is parameters of information of the periodicity (height of sound) of expression voice signal.The cycle of voice signal is different with the teller, in addition, even same talker also can change in time owing to the modulation in tone in when speech etc.By fixing index Iac is included among the coded sound signal M, make the cycle of voice signal always be fixed as a certain constant value, just become the sound of the constant robot of the height of sound by the voice signal of the code exciting lnear predict decoder (referring to Fig. 2) of the 1st embodiment regeneration.
For the telephone set of band absence recording function, also require to tackle unwanted telephone call.As a method that satisfies this requirement, the voice signal that makes callee's information voice signal become robot is effective.Therefore, the selection operation button of this pattern is set, when having operated this selection operation button, controlling the controller (CPU) of whole telephone set just will indicate the tone control signal con1 of the state of a control of tone to supply with exponential transform device 120, the fixing index Iac that will have nothing to do with the index Ia of optimal self-adaptive excitation vectors ea is included among the coded sound signal M and stores the certain basically voice signal of just exportable tone when regenerating.
Code exciting lnear predict coder according to above-mentioned the 2nd embodiment, also can obtain the effect identical with the code exciting lnear predict coder of the 1st embodiment, in addition, when regenerating, can also obtain suitably to form the effect of the certain basically voice signal of tone.
(D) the 2nd embodiment of code exciting lnear predict decoder
Fig. 4 is the 2nd embodiment of code exciting lnear predict decoder of the present invention.In Fig. 4, the part identical and corresponding with Fig. 2 is marked with identical symbol.The code exciting lnear predict decoder of the 2nd embodiment is so that be prerequisite with the code exciting lnear predict coder of above-mentioned the 1st embodiment shown in Figure 2 as encoder
By Fig. 4 and Fig. 2 more as can be known, the code exciting lnear predict decoder of the 2nd embodiment is to have increased exponential transform device 220 in the structure of the 1st embodiment.Tone control signal con1 imports this exponential transform device 220 to the index Ia of the optimal self-adaptive excitation vectors ea that separates with memory interface 201 from coded sound signal M.When tone control signal con1 indicates the non-control state of tone, exponential transform device 220 just makes the index Ia of optimal self-adaptive excitation vectors ea directly by supplying with self adaptation code table 204 and transform vector device 209, when the state of a control of tone control signal con1 indication tone, just take place with the irrelevant fixing index Iac of the index Ia of optimal self-adaptive excitation vectors ea and supply with self adaptation code table 204 and transform vector device 209.
Therefore, when tone control signal con1 indicates the non-control state of tone, the code exciting lnear predict decoder of the 2nd embodiment is deciphered with regard to the index Ia that uses the optimal self-adaptive excitation vectors ea that has separated, when tone control signal con1 indicated the state of a control of tone, the index Ia that just uses fixation index Iac to replace the optimal self-adaptive excitation vectors ea that has separated deciphered.
As a result, in the record of information voice signal, even do not carry out tone control, when regenerating, for the code exciting lnear predict coder of the 2nd embodiment since with above-mentioned identical reason, also can suitably export the certain basically voice signal of tone.
According to the code exciting lnear predict decoder of above-mentioned the 2nd embodiment, also can obtain the effect identical with the code exciting lnear predict decoder of the 1st embodiment, in addition, can also obtain suitably to form the effect of the certain basically voice signal of tone.
(E) the 3rd embodiment of code exciting lnear predict coder
Fig. 5 is the 3rd embodiment of code exciting lnear predict coder of the present invention.In Fig. 5, the part identical and corresponding with Fig. 1 is marked with identical symbol.
The code exciting lnear predict coder of the 3rd embodiment is so that be prerequisite with the code exciting lnear predict decoder (Fig. 2) of above-mentioned the 1st embodiment as decoder.
The code exciting lnear predict decoder of the code exciting lnear predict coder of the 3rd embodiment and the 3rd embodiment is to be conceived to select arbitrarily to regenerate the reproduction speed of sound vector.
By Fig. 5 and Fig. 1 more as can be known, the code exciting lnear predict coder of the 3rd embodiment is to have increased buffer storage 130, periodicity analysis device 131 and take out/interpolation operator 132 in the structure of the 1st embodiment.These new structures 131~132 are arranged on the input stage of original sound vector S, will from take out/the sound vector Sm of interpolation operator 132 outputs is as the sound import vector, the same encoding process of carrying out with the 1st embodiment.
Here, speed control signal con2 supplies with buffer storage 130~take out/interpolation operator 132.When this speed control signal con2 indication non-control state, buffer storage 130~take out/interpolation operator 132 is failure to actuate, and the original sound vector S is directly supplied with encoding process mechanism.On the other hand, when speed control signal con2 indication state of a control, buffer storage 130~take out/interpolation operator 132 just carries out the speed change action of voice signal.
Take out/action of interpolation operator 132 key diagram of (twitch do) between (becoming times multiplying power sf<1) when Fig. 6 is the indication of high rapid regeneration.As shown in Figure 6, according to the original sound vector S of 1 frame (320 sampled points), when asking its periodicity (pitch period) cc, be about 50 sampled points, in addition, according to asking when taking out between how many cycles (n cycle) are carried out, can obtain the result of 2 cycles (n=2) by becoming hits di that multiplying power sf determines.Therefore, under the situation of present embodiment, as shown in Figure 6,, take out the sampled point in 2 cycles from the beginning part of frame.So the hits of 1 frame just lacks than the hits (320) of appointment, therefore, after taking out the sampled point after the processing and the sampled point that next frame carries out taking out between same processing formed the sound vector of 1 frame again according to this, supply with encoding process mechanism.
When being the indication of low rapid regeneration, takes out between (becoming times multiplying power sf>1) Fig. 7/key diagram of the action (interpolation action) of interpolation operator 132.As shown in Figure 7, when asking its periodicity (pitch period) cc, be about 80 sampled points, in addition according to the original sound vector S of 1 frame (320 sampled points), according to asking how many cycles (n cycle) when carrying out interpolation, can obtain the result of cycle (n=2) by becoming hits di that multiplying power sf determines.Therefore, under the situation of present embodiment, as shown in Figure 7, each carries out interpolation repeatedly 2 times to the sampled point in cycle (1) of beginning one side of frame and the sampled point in the 2nd cycle (2).So, the hits of the 1 frame just hits (320) than appointment is many, therefore, 320 sampled points in the sampling row after this interpolation processing are supplied with encoding process mechanism as the sampled point of 1 frame, simultaneously, carry out supplying with encoding process mechanism after sampled point that same interpolation handles forms the sound vector of 1 frame again according to remaining sampled point with to next frame.
Telephone set for band absence recording function as described above, has the requirement that tackles the mischief phone.In addition, for the user who has a lot of phones to call in (callee's) situation, caller's absence recording information is also many, so, wish high rapid regeneration sometimes.As a method that satisfies this requirement, it is effective that the reproduction speed of callee and caller's information voice signal is changed from common speed.Therefore, be provided with the selection operation button of this pattern, when having operated this selection operation button, the controller (CPU) of controlling whole telephone set just will be indicated the change multiplying power sf of the speed control signal con2 of state of a control of reproduction speed and indication to supply with buffer storage 130~take out/interpolation operator 132 and be encoded so that coding stage (record stage) reproduction speed different with common speed.
Code exciting lnear predict coder according to above-mentioned the 3rd embodiment, also can obtain the effect identical with the code exciting lnear predict coder of the 1st embodiment, and, when regenerating, can also obtain suitably to form the effect of voice signal with reproduction speed corresponding with user's indication.
Owing to behind analytical cycle, carry out interpolation or take out processing, so, carry out interpolation or take out processing also can keep the regenerating continuity of sound, in addition, can also keep tone.
(F) the 3rd embodiment of code exciting lnear predict decoder
Fig. 8 is the 3rd embodiment of code exciting lnear predict decoder of the present invention.In Fig. 8, the part identical and corresponding with Fig. 2 is marked with identical symbol.The code exciting lnear predict decoder of the 3rd embodiment is so that be prerequisite with the code exciting lnear predict coder of above-mentioned the 1st embodiment shown in Figure 1 as encoder.
The code exciting lnear predict decoder of the 3rd embodiment also is that the common speed that the reproduction speed and sound import of voice signal itself are had is different, still, is not to carry out the change of reproduction speed in encoder one side, but is undertaken by the processing of decoder one side.
By Fig. 8 and Fig. 2 more as can be known, the code exciting lnear predict decoder of the 3rd embodiment is to have increased buffer storage 230, periodicity analysis device 231 and take out/interpolation operator 232 between the adder 212 of the 1st embodiment and composite filter 213, in addition, also the appointment hits hits in addition with 1 frame is corresponding for composite filter 213 and postfilter 214.Therefore, the code exciting lnear predict decoder of processing before the adder 212 and the 1st embodiment is identical.
Here, speed control signal con2 is supplied with buffer storage 230~take out/interpolation operator 232.When this speed control signal con2 indication non-control state, buffer storage 230~take out/interpolation operator 232 is failure to actuate, and Optimum Excitation vector e is directly passed through.On the other hand, when speed control signal con2 indication state of a control, buffer storage 230~take out/interpolation operator 232 carries out speed change action.
But, take out between the code exciting lnear predict decoder of the 3rd embodiment/interpolation operator 232 so also ask between take out/vector length (hits) s1 of Optimum Excitation vector em after interpolation is handled.And, take out/interpolation operator 232 will between take out/Optimum Excitation vector em after interpolation is handled is defeated by composite filter 213, simultaneously, vector length s1 is defeated by composite filter 213 and postfilter 214.
Code exciting lnear predict decoder according to above-mentioned the 3rd embodiment, also can obtain the effect identical with the code exciting lnear predict decoder of the 1st embodiment, and, can also obtain suitably to form the effect of regeneration voice signal with reproduction speed corresponding with user's indication.
Here, owing to take out between analytical cycle after, carrying out or interpolation is handled, so, even take out between carrying out or interpolation processing can also keep the regenerating continuity of sound, in addition, can also keep tone.
In addition, owing to be to smoke between carrying out in the stage of Optimum Excitation vector or interpolation is handled, so, more natural regeneration synthetic video signal can be obtained.That is, by take out/interpolation handles the influence that the causes filtering by composite filter 213 and postfilter 214 and relaxes, and can obtain more natural regeneration voice signal.Therefore, also can consider between the output stage of postfilter 214 carries out, to take out/interpolation handles, still, even carrying out that periodicity analysis is laggard to be taken out in the ranks/interpolation handles, and its influence enters the degree of output sound signal also greater than present embodiment.
(G) the 4th embodiment of code exciting lnear predict decoder
Fig. 9 is the 4th embodiment of code exciting lnear predict decoder of the present invention.In Fig. 9, the part identical and corresponding with Fig. 2 is marked with identical symbol.The code exciting lnear predict decoder of the 4th embodiment is so that be prerequisite with the code exciting lnear predict coder of above-mentioned the 1st embodiment shown in Figure 1 as encoder.
The code exciting lnear predict coder of the 1st embodiment shown in Figure 1 as described above, is should be able to be with a large amount of information stores in the IC memory and have a low coding rate during coding.How many coding rates reduces, and the coding distortion enters in the regeneration voice signal just what are arranged, and this is inevitable.Know that experimentally because the coding distortion, the noise composition in the regeneration voice signal has the tendency that becomes the microphonic noiseization.The code exciting lnear predict decoder of the 4th embodiment is exactly in order to solve the problem that the microphonic noiseization appears in the noise composition of regeneration in the voice signal.
By Fig. 9 and Fig. 2 more as can be known, the code exciting lnear predict decoder of the 4th embodiment is to have increased noise generator 140 and adder 141 in the structure of the 1st embodiment.
White noise nz takes place with the value of frame power P in noise generator 140 accordingly.In addition, the certain noise that changes of maintenance that has nothing to do with frame power taking place or catch in advance after background noise is stored noise to take place again, can constitute other embodiment.Adder 141 is carried out add operation with the regeneration sound vector of this noise nz and postfilter 214 outputs, and the regeneration sound vector Sp after the add operation is exported to the outside from lead-out terminal 215.
Here, even the microphonic noiseization appears in the noise composition from the regeneration sound vector of postfilter 214 outputs, by adding the white noise of noise generator 140, just can make the noise composition of the regeneration sound vector Sp of adder 141 outputs present white noiseization, thereby microphonic noise composition becomes not obvious, approaches the noise composition of nature.
Code exciting lnear predict decoder according to above-mentioned the 4th embodiment, also can obtain the effect identical with the code exciting lnear predict decoder of the 1st embodiment, and, even make background noise etc. be modulated, be changed to feel the degree that influence is listened to by coding/decoding, owing to add the noise that generates artificially of the suitable size of appointment, so, also can shelter the part that influence is listened to, thereby can obtain more natural regeneration voice signal.
(H) other embodiment
In the various embodiments described above, show according to the what is called that obtains the sound channel coefficient of analysis according to the original sound vector situation of formula code-excited linear prediction mode forward, but, for the feature structure of the 1st embodiment, the 3rd embodiment and the 4th embodiment, the what is called that can be applied to obtain the sound channel coefficient of analysis according to the sound vector according to part regeneration is the situation of formula code-excited linear prediction mode backward.
In the various embodiments described above, recurring structure as pumping signal (excitation vectors), have self adaptation code table, noise code table, pulse code table and gain code table, but, for the 2nd embodiment~the 4th embodiment, the recurring structure of pumping signal (excitation vectors) is not limited thereto, so long as have self adaptation code table and noise code table at least, just can use.
The various embodiments described above all are to be conceived to be applied in the structure of information record regenerating of telephone set of band absence recording function, and still, its purposes is not limited thereto, and can be applied to the transmission system of narrow sense.
According to the 1st code exciting lnear predict coder of the present invention or the 2nd code exciting lnear predict decoder of the present invention, be that the sound channel predictive coefficient that order is included in the coded sound signal is the LSP coefficient, select to use noise code table and pulse feature code table, select the frequency of reflection original sound signal in the pumping signal at this, to the adaptive excitation signal with select pumping signal to carry out gain controlling, so, even under low coding rate, also can obtain high-quality regeneration voice signal.
According to the 3rd code exciting lnear predict coder of the present invention or the 4th code exciting lnear predict decoder of the present invention, owing to be provided with the exponential transform device, when having indicated the tone control pattern, the exponential transform of adaptive excitation signal is fixing adaptive excitation signal index, so, can change as required regeneration voice signal tone.
According to the 5th code exciting lnear predict coder of the present invention or the 6th code exciting lnear predict decoder of the present invention, take out between according to the change multiplying power of indication and the periodicity that has of process object signal the process object signal being carried out or reproduction speed controller that interpolation is handled owing to be provided with, so, can change as required regeneration voice signal speed.
According to the 7th code exciting lnear predict decoder of the present invention, be provided with the white noise adder that in the regeneration voice signal, adds white noise owing to form the back level of device at the regeneration voice signal, so, though the microphonic noise takes place in the noise composition of regeneration voice signal easily under low coding rate, but, the microphonic noise is flooded in the white noise just not obvious, thereby can obtain the regeneration voice signal of nature.
Claims (7)
1. code exciting lnear predict coder, it is characterized in that: have channel information maker, sound power quantizer, self adaptation code table, noise code table, pulse code table, constant excitation signal selector, frequency characteristic operator, gain code table, adder, Optimum Excitation signal searcher of Fig and coded sound signal and form device, the channel information maker obtains the LPC coefficient, quantizes after it is transformed to the LSP coefficient according to the synthetic video signal of original sound signal or local regeneration, the LSP coefficient that will quantize simultaneously carries out inverse quantization, reverts to the LPC coefficient; The sound power quantizer calculates the power of original sound signal and quantizes and the inverse quantization processing; The self adaptation code table is exported the adaptive excitation signal of renewal adaptively; Noise code table output noise pumping signal; Pulse code table output pulse feature pumping signal; The constant excitation signal selector is selected the pumping signal of noise code table and pulse code table; The frequency characteristic operator is transformed to the frequency characteristic that has with the original sound signal similar with the noise pumping signal or the pulse feature pumping signal of the output of constant excitation signal selector; Gain code table output with from the adaptive excitation signal of self adaptation code table output with from the noise pumping signal of frequency characteristic operator output or pulse feature pumping signal excitation gain corresponding, that determine by the inverse quantization power of sound power quantizer output; What adder will have been carried out gain controlling carries out add operation from the adaptive excitation signal of self adaptation code table output with from the noise pumping signal or the pulse feature pumping signal of the output of frequency characteristic operator, forms final pumping signal; The Optimum Excitation signal searcher of Fig is according to the final pumping signal of exporting from the LPC coefficient and the adder of the output of channel information maker, determine to use Optimum Excitation signal, also be to use the output drive signal of which code table in noise code table and the pulse code table from self adaptation code table, noise code table, pulse code table and the output of gain code table; Code signal forms device when having determined the Optimum Excitation signal, forms coded sound signal according to the index of various code tables, the selection mode index of constant excitation signal selector, the quantized power of sound power quantizer and the quantification LSP coefficient of channel information maker.
2. a code exciting lnear predict decoder is characterized in that: have coded sound signal separator, channel information regenerator, sound power inverse DCT, self adaptation code table, noise code table, pulse code table, constant excitation signal selector, frequency characteristic operator, gain code table, adder and regeneration voice signal and form device.The coded sound signal separator is separated into various information with coded sound signal; The quantification LSP coefficient that the channel information regenerator will separate carries out inverse quantization to be handled, and reverts to the LPC coefficient; The sound power inverse DCT carries out inverse quantization with the quantized power of separating to be handled; Output of self adaptation code table and the corresponding adaptive excitation signal of adaptive excitation signal index that separates; The noise code table is exported the noise pumping signal corresponding with this index when having separated noise pumping signal index; The pulse code table is exported the pulse feature pumping signal corresponding with this index when having separated pulse feature pumping signal index; The constant excitation signal selector is selected the pumping signal of noise code table and pulse code table according to the selection mode index that separates; The frequency characteristic operator will be transformed to the frequency characteristic that has with the original sound signal similar from noise pumping signal or the pulse feature pumping signal that this constant excitation signal is far selected device output; Gain code table output with from the adaptive excitation signal of self adaptation code table output with from the noise pumping signal of frequency characteristic operator output or pulse feature pumping signal corresponding, by the excitation gain of determining from the inverse quantization power of sound power inverse DCT output and the excitation gain that separates; Adder will form final pumping signal from carrying out add operation through the adaptive excitation signal of the self adaptation code table of gain controlling output with from the noise pumping signal or the pulse feature pumping signal of the output of frequency characteristic operator; The regeneration voice signal forms device according to forming the regeneration voice signal from the LPC coefficient of channel information regenerator output with from the final pumping signal that adder is exported.
3. code exciting lnear predict coder, it is characterized in that: in the code exciting lnear predict coder that has the self adaptation code table at least, has the exponential transform device, when having indicated the tone control pattern, this exponential transform device will be fixing adaptive excitation signal index according to the adaptive excitation signal exponential transform that the self adaptation code table is determined, and supply with coded sound signal formation device, so that contain fixing adaptive excitation signal in the coded sound signal.
4. code exciting lnear predict decoder, it is characterized in that: in having the code exciting lnear predict decoder of self adaptation code table at least, has the exponential transform device, when having indicated the tone control pattern, this exponential transform device will be transformed to fixing adaptive excitation signal index and supply with the self adaptation code table according to the definite adaptive excitation signal index of self adaptation code table that obtains after coded sound signal is separated.
5. code exciting lnear predict coder, it is characterized in that: in the code exciting lnear predict coder of the original sound signal of input being encoded according to the code-excited linear prediction mode, has the reproduction speed controller, when having indicated the reproduction speed control model, this reproduction speed controller is according to the change multiplying power and the periodicity that has of original sound signal of indication, takes out between the original sound signal is carried out or encode processor is handled and supplied with to interpolation.
6. code exciting lnear predict decoder, it is characterized in that: have the pumping signal regenerator that forms pumping signal according to the coded sound signal of input, form the channel information regenerator of sound channel predictive coefficient and form in the code exciting lnear predict decoder of device according to the coded sound signal of input according to the regeneration voice signal that the sound channel predictive coefficient of the pumping signal of pumping signal regenerator and sound channel information regeneration device forms the regeneration voice signal, has the reproduction speed controller, when having indicated the reproduction speed control model, take out between this reproduction speed controller carries out the pumping signal of pumping signal regenerator according to the change multiplying power of indication and the periodicity that has of pumping signal or interpolation is handled, and supply with the regeneration voice signal and form device.
7. code exciting lnear predict decoder, it is characterized in that: have the pumping signal regenerator that forms pumping signal according to the coded sound signal of input, form the channel information regenerator of sound channel predictive coefficient and form in the code exciting lnear predict decoder of device according to the regeneration voice signal that the sound channel predictive coefficient of the pumping signal of pumping signal regenerator and sound channel information regeneration device forms the regeneration voice signal according to the coded sound signal of input, the back level that forms device at the regeneration voice signal is provided with the white noise adder of adding white noise in the regeneration voice signal.
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Families Citing this family (44)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US5717823A (en) * | 1994-04-14 | 1998-02-10 | Lucent Technologies Inc. | Speech-rate modification for linear-prediction based analysis-by-synthesis speech coders |
US5774846A (en) * | 1994-12-19 | 1998-06-30 | Matsushita Electric Industrial Co., Ltd. | Speech coding apparatus, linear prediction coefficient analyzing apparatus and noise reducing apparatus |
SE506379C3 (en) * | 1995-03-22 | 1998-01-19 | Ericsson Telefon Ab L M | Lpc speech encoder with combined excitation |
JP3092652B2 (en) * | 1996-06-10 | 2000-09-25 | 日本電気株式会社 | Audio playback device |
EP0883107B9 (en) | 1996-11-07 | 2005-01-26 | Matsushita Electric Industrial Co., Ltd | Sound source vector generator, voice encoder, and voice decoder |
JP3206497B2 (en) * | 1997-06-16 | 2001-09-10 | 日本電気株式会社 | Signal Generation Adaptive Codebook Using Index |
EP1734512B1 (en) | 1997-10-22 | 2015-09-09 | Godo Kaisha IP Bridge 1 | CELP encoder and a method of CELP encoding |
US6092040A (en) * | 1997-11-21 | 2000-07-18 | Voran; Stephen | Audio signal time offset estimation algorithm and measuring normalizing block algorithms for the perceptually-consistent comparison of speech signals |
US6385576B2 (en) * | 1997-12-24 | 2002-05-07 | Kabushiki Kaisha Toshiba | Speech encoding/decoding method using reduced subframe pulse positions having density related to pitch |
CA2636552C (en) * | 1997-12-24 | 2011-03-01 | Mitsubishi Denki Kabushiki Kaisha | A method for speech coding, method for speech decoding and their apparatuses |
KR100249235B1 (en) * | 1997-12-31 | 2000-03-15 | 구자홍 | Hdtv video decoder |
US5963897A (en) * | 1998-02-27 | 1999-10-05 | Lernout & Hauspie Speech Products N.V. | Apparatus and method for hybrid excited linear prediction speech encoding |
US6104992A (en) * | 1998-08-24 | 2000-08-15 | Conexant Systems, Inc. | Adaptive gain reduction to produce fixed codebook target signal |
US7117146B2 (en) * | 1998-08-24 | 2006-10-03 | Mindspeed Technologies, Inc. | System for improved use of pitch enhancement with subcodebooks |
US7072832B1 (en) * | 1998-08-24 | 2006-07-04 | Mindspeed Technologies, Inc. | System for speech encoding having an adaptive encoding arrangement |
US6691084B2 (en) | 1998-12-21 | 2004-02-10 | Qualcomm Incorporated | Multiple mode variable rate speech coding |
US6311154B1 (en) * | 1998-12-30 | 2001-10-30 | Nokia Mobile Phones Limited | Adaptive windows for analysis-by-synthesis CELP-type speech coding |
US6449313B1 (en) * | 1999-04-28 | 2002-09-10 | Lucent Technologies Inc. | Shaped fixed codebook search for celp speech coding |
US6728344B1 (en) * | 1999-07-16 | 2004-04-27 | Agere Systems Inc. | Efficient compression of VROM messages for telephone answering devices |
JP3365360B2 (en) * | 1999-07-28 | 2003-01-08 | 日本電気株式会社 | Audio signal decoding method, audio signal encoding / decoding method and apparatus therefor |
US6452517B1 (en) * | 1999-08-03 | 2002-09-17 | Dsp Group Ltd. | DSP for two clock cycle codebook search |
US6959274B1 (en) | 1999-09-22 | 2005-10-25 | Mindspeed Technologies, Inc. | Fixed rate speech compression system and method |
US7133823B2 (en) * | 2000-09-15 | 2006-11-07 | Mindspeed Technologies, Inc. | System for an adaptive excitation pattern for speech coding |
US6678651B2 (en) * | 2000-09-15 | 2004-01-13 | Mindspeed Technologies, Inc. | Short-term enhancement in CELP speech coding |
JP3566220B2 (en) * | 2001-03-09 | 2004-09-15 | 三菱電機株式会社 | Speech coding apparatus, speech coding method, speech decoding apparatus, and speech decoding method |
US6912495B2 (en) * | 2001-11-20 | 2005-06-28 | Digital Voice Systems, Inc. | Speech model and analysis, synthesis, and quantization methods |
JP4433668B2 (en) * | 2002-10-31 | 2010-03-17 | 日本電気株式会社 | Bandwidth expansion apparatus and method |
US20040102975A1 (en) * | 2002-11-26 | 2004-05-27 | International Business Machines Corporation | Method and apparatus for masking unnatural phenomena in synthetic speech using a simulated environmental effect |
WO2004090870A1 (en) | 2003-04-04 | 2004-10-21 | Kabushiki Kaisha Toshiba | Method and apparatus for encoding or decoding wide-band audio |
KR100651712B1 (en) * | 2003-07-10 | 2006-11-30 | 학교법인연세대학교 | Wideband speech coder and method thereof, and Wideband speech decoder and method thereof |
US8332216B2 (en) * | 2006-01-12 | 2012-12-11 | Stmicroelectronics Asia Pacific Pte., Ltd. | System and method for low power stereo perceptual audio coding using adaptive masking threshold |
JP4525694B2 (en) * | 2007-03-27 | 2010-08-18 | パナソニック株式会社 | Speech encoding device |
JP4525693B2 (en) * | 2007-03-27 | 2010-08-18 | パナソニック株式会社 | Speech coding apparatus and speech decoding apparatus |
US9343079B2 (en) * | 2007-06-15 | 2016-05-17 | Alon Konchitsky | Receiver intelligibility enhancement system |
EP2269188B1 (en) | 2008-03-14 | 2014-06-11 | Dolby Laboratories Licensing Corporation | Multimode coding of speech-like and non-speech-like signals |
EP2313986A1 (en) * | 2008-08-13 | 2011-04-27 | Nokia Siemens Networks Oy | Method of generating a codebook |
JP5299631B2 (en) * | 2009-05-13 | 2013-09-25 | 日本電気株式会社 | Speech decoding apparatus and speech processing method thereof |
JP5287502B2 (en) * | 2009-05-26 | 2013-09-11 | 日本電気株式会社 | Speech decoding apparatus and method |
CN101834586A (en) * | 2010-04-21 | 2010-09-15 | 四川和芯微电子股份有限公司 | Random signal generating circuit and method |
DK2774145T3 (en) * | 2011-11-03 | 2020-07-20 | Voiceage Evs Llc | IMPROVING NON-SPEECH CONTENT FOR LOW SPEED CELP DECODERS |
EP3011561B1 (en) | 2013-06-21 | 2017-05-03 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Apparatus and method for improved signal fade out in different domains during error concealment |
EP2980799A1 (en) * | 2014-07-28 | 2016-02-03 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Apparatus and method for processing an audio signal using a harmonic post-filter |
CN105007094B (en) * | 2015-07-16 | 2017-05-31 | 北京中宸泓昌科技有限公司 | A kind of exponent pair spread spectrum coding coding/decoding method |
EP3704863B1 (en) * | 2017-11-02 | 2022-01-26 | Bose Corporation | Low latency audio distribution |
Citations (2)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
CN1047157A (en) * | 1989-05-11 | 1990-11-21 | 艾利森电话股份有限公司 | The localization method of linear predict voice coding driving pulse |
CN1051100A (en) * | 1989-10-17 | 1991-05-01 | 莫托罗拉公司 | The linear predictive coding voice synthesizer that has the adaptive pitch prefilter |
Family Cites Families (23)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
JPS5650398A (en) * | 1979-10-01 | 1981-05-07 | Hitachi Ltd | Sound synthesizer |
US4624012A (en) * | 1982-05-06 | 1986-11-18 | Texas Instruments Incorporated | Method and apparatus for converting voice characteristics of synthesized speech |
FR2530101A1 (en) * | 1982-07-06 | 1984-01-13 | Thomson Brandt | Process and system for encrypted transmission of a signal, especially of audio frequency |
US4709390A (en) * | 1984-05-04 | 1987-11-24 | American Telephone And Telegraph Company, At&T Bell Laboratories | Speech message code modifying arrangement |
JP2884163B2 (en) * | 1987-02-20 | 1999-04-19 | 富士通株式会社 | Coded transmission device |
US4868867A (en) * | 1987-04-06 | 1989-09-19 | Voicecraft Inc. | Vector excitation speech or audio coder for transmission or storage |
EP0287741B1 (en) * | 1987-04-22 | 1993-03-31 | International Business Machines Corporation | Process for varying speech speed and device for implementing said process |
DE68922134T2 (en) * | 1988-05-20 | 1995-11-30 | Nec Corp | Coded speech transmission system with codebooks for synthesizing low amplitude components. |
EP0427953B1 (en) * | 1989-10-06 | 1996-01-17 | Matsushita Electric Industrial Co., Ltd. | Apparatus and method for speech rate modification |
JPH0451199A (en) * | 1990-06-18 | 1992-02-19 | Fujitsu Ltd | Sound encoding/decoding system |
US5138661A (en) * | 1990-11-13 | 1992-08-11 | General Electric Company | Linear predictive codeword excited speech synthesizer |
US5293449A (en) * | 1990-11-23 | 1994-03-08 | Comsat Corporation | Analysis-by-synthesis 2,4 kbps linear predictive speech codec |
US5537509A (en) * | 1990-12-06 | 1996-07-16 | Hughes Electronics | Comfort noise generation for digital communication systems |
US5195137A (en) * | 1991-01-28 | 1993-03-16 | At&T Bell Laboratories | Method of and apparatus for generating auxiliary information for expediting sparse codebook search |
JP2776050B2 (en) * | 1991-02-26 | 1998-07-16 | 日本電気株式会社 | Audio coding method |
JP2661391B2 (en) * | 1991-03-01 | 1997-10-08 | ヤマハ株式会社 | Music signal processor |
US5396576A (en) * | 1991-05-22 | 1995-03-07 | Nippon Telegraph And Telephone Corporation | Speech coding and decoding methods using adaptive and random code books |
US5175769A (en) * | 1991-07-23 | 1992-12-29 | Rolm Systems | Method for time-scale modification of signals |
EP0527527B1 (en) * | 1991-08-09 | 1999-01-20 | Koninklijke Philips Electronics N.V. | Method and apparatus for manipulating pitch and duration of a physical audio signal |
US5305420A (en) * | 1991-09-25 | 1994-04-19 | Nippon Hoso Kyokai | Method and apparatus for hearing assistance with speech speed control function |
WO1993018505A1 (en) * | 1992-03-02 | 1993-09-16 | The Walt Disney Company | Voice transformation system |
US5765127A (en) * | 1992-03-18 | 1998-06-09 | Sony Corp | High efficiency encoding method |
SG43128A1 (en) * | 1993-06-10 | 1997-10-17 | Oki Electric Ind Co Ltd | Code excitation linear predictive (celp) encoder and decoder |
-
1994
- 1994-11-22 JP JP28765494A patent/JP3328080B2/en not_active Expired - Fee Related
-
1995
- 1995-10-13 KR KR1019950035415A patent/KR100272477B1/en not_active IP Right Cessation
- 1995-11-14 US US08/557,809 patent/US5752223A/en not_active Expired - Lifetime
- 1995-11-16 EP EP01108216A patent/EP1160771A1/en not_active Withdrawn
- 1995-11-16 DE DE69527410T patent/DE69527410T2/en not_active Expired - Fee Related
- 1995-11-16 EP EP95118092A patent/EP0714089B1/en not_active Expired - Lifetime
- 1995-11-17 CN CN95119729A patent/CN1055585C/en not_active Expired - Fee Related
Patent Citations (2)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
CN1047157A (en) * | 1989-05-11 | 1990-11-21 | 艾利森电话股份有限公司 | The localization method of linear predict voice coding driving pulse |
CN1051100A (en) * | 1989-10-17 | 1991-05-01 | 莫托罗拉公司 | The linear predictive coding voice synthesizer that has the adaptive pitch prefilter |
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KR100272477B1 (en) | 2000-11-15 |
CN1132423A (en) | 1996-10-02 |
KR960019069A (en) | 1996-06-17 |
JPH08146998A (en) | 1996-06-07 |
JP3328080B2 (en) | 2002-09-24 |
EP1160771A1 (en) | 2001-12-05 |
EP0714089B1 (en) | 2002-07-17 |
DE69527410T2 (en) | 2003-08-21 |
EP0714089A3 (en) | 1998-07-15 |
EP0714089A2 (en) | 1996-05-29 |
US5752223A (en) | 1998-05-12 |
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